KR101894708B1 - Speaker damage prevention in adaptive noise-canceling personal audio devices - Google Patents

Abstract

A personal audio device, such as a cordless phone, includes a noise cancellation circuit that generates a noise-proof signal from a reference microphone signal and injects a noise-proof signal to the speaker or other transducer output, It causes. The processing circuit monitors the level of the anti-noise signal and determines whether the anti-noise signal can cause damage to the transducer, and the generation of the anti-noise signal to prevent damage to the transducer .

Description

≪ Desc / Clms Page number 1 > SPEAKER DAMAGE PREVENTION IN ADAPTIVE NOISE-CANCELING PERSONAL AUDIO DEVICES IN PERSONAL AUDIO DEVICES WITH ADAPTIVE NOISE ERASE

The present invention relates to personal audio devices, such as wireless phones, including noise cancellation, and more particularly to a personal audio device in which damage to an output transducer is prevented while still providing adaptive noise cancellation.

Other consumer audio devices, such as mobile / cellular telephones, cordless telephones such as cordless phones, and mp3 players, are widely deployed and used. The performance of these devices with regard to malleability can be improved by measuring ambient acoustic events using a microphone and then using signal processing to provide noise cancellation to erase ambient acoustic events by inserting a noise-preventing signal at the device output.

It is desirable to adapt noise cancellation to account for these environmental changes, since the acoustic environment around personal audio devices such as wireless telephones may vary dramatically depending on the noise sources present and the location of the device itself. However, the adaptive noise cancellation circuit can be complex, can consume additional power, and produce undesirable results under certain circumstances.

Therefore, it would be desirable to provide a personal audio device that includes a cordless phone that provides noise cancellation in a variety of environments.

The above-mentioned object of providing a personal audio device, which provides noise cancellation in various environments, is achieved as a personal audio device, method of operation, and integrated circuit.

The personal audio device includes a housing and is equipped with a transducer for reproducing an audio signal in the housing, and the audio signal is supplied to the listener in a noise-canceling manner to cancel the influence of ambient audio sounds in the source audio to be reproduced and the acoustic output of the transducer, Prevention signal. A reference microphone is mounted in the housing to provide a reference microphone signal indicative of ambient audio sounds. The personal audio device includes an adaptive noise-canceling (ANC) processing circuit within the housing for adaptively generating a noise-preventing signal from the reference microphone signal such that the noise-preventing signal causes a substantial erasure of ambient audio sounds. The ANC processing circuitry monitors the level of the anti-noise signal to determine if the anti-noise signal can cause damage to the transducer and adjusts the generation of the anti-noise signal to prevent damage to the transducer. The integrated circuit includes a processing circuit that performs such monitoring and adjustment, and the method is a method of operation of the integrated circuit.

The foregoing and other objects, features, and advantages of the present invention will become apparent from the following more particular description of preferred embodiments of the invention, as illustrated in the accompanying drawings.

1 illustrates a wireless telephone 10 in accordance with an embodiment of the present invention.
2 is a block diagram of circuits within a wireless telephone 10 in accordance with one embodiment of the present invention.
3 is a block diagram illustrating signal processing circuits and functional blocks within the ANC circuit 30 of the CODEC integrated circuit 20 of FIG. 2 according to one embodiment of the present invention.
4 is a block diagram illustrating details of the speaker damage prevention circuit 60 of FIG. 3 in accordance with one embodiment of the present invention.
5 is a block diagram illustrating signal processing circuitry and functional blocks within an integrated circuit in accordance with one embodiment of the present invention.

The present invention includes noise cancellation techniques and circuits that may be implemented in a personal audio device such as a wireless telephone. The personal audio device includes adaptive noise cancellation (ANC) circuitry that measures the ambient acoustic environment and generates an adaptive signal that is inserted into the speaker (or other transducer) output to clear the ambient acoustic events. The ANC circuit monitors the level of the anti-noise signal to determine if damage to the speaker or other transducer is urgent and adjusts the anti-noise signal if speaker damage can occur.

Referring now to FIG. 1, a wireless telephone 10 shown in accordance with an embodiment of the present invention is shown proximate the human ear 5. The illustrated wireless telephone 10 is an example of a device in which techniques according to embodiments of the present invention may be implemented, but in the illustrated wireless telephone 10, or in the circuits illustrated in the following descriptions, Elements or configurations are not required to practice the invention set forth in the claims. The wireless telephone 10 includes a transducer, such as a speaker (SPKR), that reproduces far-away audio received by the wireless telephone 10 with other local audio events, which may include ring tones, Program material, infusion of near-end speech (i.e., the user's voice of the wireless telephone 10) to provide balanced conversation recognition, and other audio that requires playback by the wireless telephone 10 And other audio may include sources from web-pages or other network communications received by wireless telephone 10 and audio indications such as battery low and other system event notifications. A near-speech microphone NS is provided to capture the near-end voice and a near-end voice is transmitted from the wireless telephone 10 to the other conversation participant (s).

The wireless telephone 10 includes adaptive noise cancellation (ANC) circuits and features that inject noise-preventing signals into the speaker SPKR to improve the versatility of distant voice and other audio reproduced by the speaker SPKR do. The reference microphone R is provided for measuring the ambient acoustic environment and is located away from the typical position of the user's mouth such that the near-end voice in the signal generated by the reference microphone R is minimized. The third microphone, error microphone E, is a measure of ambient audio combined with audio reproduced by a speaker (SPKR) close to ear 5 when cordless telephone 10 is near ear 5 To < / RTI > further improve ANC operation. Exemplary circuitry 14 within wireless telephone 10 is configured to receive signals from a reference microphone R, a near-voiced microphone NS and an error microphone E and to receive a radio frequency (RF) And an audio CODEC integrated circuit 20 that interfaces with other integrated circuits, such as the integrated circuit 12. In other embodiments of the present invention, the circuits and techniques disclosed herein may be implemented within a single integrated circuit, including control circuits and other functions for implementing the entirety of a personal audio device, such as an MP3 player integrated circuit on a chip Lt; / RTI >

In general, the ANC techniques of the present invention measure ambient acoustic events (as opposed to the output and / or near-end speech of the speaker SPKR) that affect the reference microphone R, and also to the error microphone E By measuring the same peripheral acoustic events that are affecting, the ANC processing circuits of the illustrated cordless telephone 10 are able to detect the noise-preventing signal generated from the output of the reference microphone R from the error microphone E to the amplitude of the ambient acoustic event To be minimized. Since the acoustic path P (z) extends from the reference microphone R to the error microphone E, the ANC circuits essentially have an acoustic path coupled with the movement effects of the electro-acoustic path S (z) Acoustic path S (z) is determined by the response of the audio output circuits of the CODEC IC 20 and the coupling between the speaker SPKR and the error microphone E in a particular acoustic environment, S (z) represents the acoustic / electrical transfer function of the speaker (SPKR) including the proximity and structure of the ear (5) and other physical objects when the cordless telephone is not reliably pressed on the ear (5) And are affected by human head structures that may be near the wireless telephone 10. [ Although the illustrated cordless telephone 10 includes two microphone ANC systems with a third near-vocal microphone (NS), some aspects of the present invention may include a system that does not include separate error and reference microphones, May be implemented in a cordless telephone using a near-vocal microphone (NS) to perform the function of the microphone (R). Also, for personal audio devices designed exclusively for audio playback, a near-vocal microphone (NS) will generally not be included, and near-voice signal paths within the circuits described in more detail below, Can be omitted without changing the category of "

Referring now to FIG. 2, the circuits within wireless telephone 10 are shown in block diagram form. The CODEC integrated circuit 20 includes an analog-to-digital converter (ADC) 21A for receiving a reference microphone signal and generating a digital representation ref of the reference microphone signal, a digital representation of the error microphone signal err), and an ADC 21C for receiving the near-voiced microphone signal and generating a digital representation (ns) of the near-voiced microphone signal. The CODEC integrated circuit 20 produces an output for driving the speaker SPKR from the amplifier A1 and the amplifier A1 is connected to the output of a digital-to-analog converter (DAC) 23 for receiving the output of the combiner 26 Lt; / RTI > The combiner 26 receives the audio signals from the internal audio sources 24 and the ANC circuit 30 having the same polarity as the noise in the reference microphone signal ref by convention and thus subtracted by the combiner 26. [ Combines the noise-suppression signal generated by the noise suppression unit. The combiner 26 injects a portion of the near-voice signal ns so that the user of the radiotelephone 10 hears their voice in an appropriate manner to the downlink voice ds, Is received from the RF integrated circuit 22 and is coupled by the coupler 26. [ The near-voice signal ns is also provided to the RF integrated circuit 22 and transmitted as an uplink voice to the mobile telephone service provider via the antenna ANT.

Referring now to FIG. 3, the details of the ANC circuit 30 are shown in accordance with one embodiment of the present invention. The adaptive filter 32 receives the reference microphone signal ref and adapts the transfer function W (Z) to be P (z) / S (z) under ideal circumstances to produce a noise-proof signal. Coefficients of the adaptive filter 32 are controlled by the W coefficient control block 31 and the W coefficient control block 31 compares the components of the reference microphone signal ref with the error microphone signal err, The correlation of the two signals is used to determine the response of the adaptive filter 32 that generally minimizes the error with respect to the mean square. W coefficients control block 31 compares the reference microphone signal ref shaped by the replica of the estimate of the path S (z) provided by the filter 34B with the error microphone signal err, ≪ / RTI > By transforming the reference microphone signal ref through a replica of the estimate of the response of path S (z) (SE _COPY (z)) and minimizing the difference between the final signal and the error microphone signal err, (Z) is adapted to the desired response of P (Z) / S (z) by adapting to eliminate the effect of applying the response SE _COPY (z) from the reference microphone signal ref. In addition to the error microphone signal err, the signal compared by the W-coefficient control block 31 to the output of the filter 34B is the signal of the downlink audio signal ds processed by the filter response SE (z) Includes an inverted amount, the response (SE _COPY (z)) is a duplicate. By injecting the inverted amount of the downlink audio signal ds, the adaptive filter 32 is prevented from adapting to the relatively large amount of downlink audio present in the error microphone signal err, and the path S (z ), The downlink audio signal removed from the error microphone signal err before the comparison is converted to the down-stream audio signal ds reproduced from the error microphone signal err by converting the inverse replica of the downlink audio signal ds having an estimate of the response It is necessary to match the expected form of the link audio signal ds since the electrical and acoustic path of S (z) is the path taken by the downlink audio signal ds to reach the error microphone E.

In order to implement the above, the adaptive filter 34A has coefficients controlled by the SE coefficient control block 33, and the SE coefficient control block 33 compares the downlink audio signal ds with the above- The downlink audio signal ds is filtered by the adaptive filter 34A and compared to the estimated downlink signal ds that is delivered to the error microphone E by comparing the error microphone signal err after the removal of the downlink audio signal ds, Audio and is removed from the output of the adaptive filter 34A by the combiner 36. [ The SE coefficient control block 33 correlates the actual downlink speech signal ds with the components of the downlink audio signal ds present in the error microphone signal err. The adaptive filter 34A is thereby adapted to generate a signal from the downlink audio signal ds when the downlink audio signal ds is subtracted from the error microphone signal err, And contains the content of the error microphone signal err which is not caused. The event detection and control logic 38 performs various actions in response to various events in accordance with various embodiments of the present invention, as described in more detail below.

Since the adaptive filter 32 has a wide gain at various frequencies depending on the environment in which the W coefficient control 31 adapts the response of the adaptive filter 32 the noise- The prevention signal may assume a high amplitude, which can cause damage to the speaker (SPKR), especially at low frequencies where the speaker (SPKR) has a poor acoustic response. Since the high amplitudes will attempt to cancel any low frequency peripheral events by raising the gain of the adaptive filter 32 in these frequency bands regardless of the frequency response of the speaker SPKR, Occurs. Moreover, low frequency signal components can stimulate resonances that further damage the speaker (SPKR) rather than higher frequency components. Therefore, the speaker damage prevention circuit 60 is included in the ANC circuit 20 to prevent damage to the speaker SPKR by processing the noise-prevention signal.

Referring now to FIG. 4, there is shown details of a speaker damage prevention circuit 60 according to an embodiment of the present invention. The input signal in is received from the output of the adaptive filter 32 and the multiplier 66A applies various attenuation values atten1 determined by the signal level detector 64A, Gt; of the input signal < RTI ID = 0.0 > (in) < / RTI > The low pass filter 62 removes high frequency components from the input signal in, such as frequencies above 500 Hz, so that the attenuation value atten1 is equal to the energy in the input signal in, which lies within a frequency range of less than 500 Hz. Almost entirely. The multiplier 66A provides a gain control block that does not filter the input signal in and adjusts the level of the input signal in, i.e., changes only the overall gain without changing the spectrum of the input signal in . Another multiplier 66B provides a second gain control cell that adjusts the level of the output of the first multiplier 66A in accordance with the value chain atten2 and the value chain atten2 is provided to the second signal level detector 64B Is determined from the unfiltered output of the first multiplier 66A. In the illustrated embodiment, the signal level detectors 64A and 66B are threshold detectors, i.e., the attenuation values atten1 and atten2 correspond to the corresponding signal levels reaching the inputs of the signal level detectors 64A and 66B Once a predetermined threshold has been exceeded. Also, the change in the attenuation values atten1 and atten2 for the signal levels is the manner in which the infinite compression ratio is applied, i.e. the attenuation values atten1 and atten2 are set such that the corresponding signal levels do not exceed their corresponding thresholds Change to ensure that. Therefore, the low-pass filter 62, the signal level detector 64A and the multiplier 66A form the first soft limiter and the signal level detector 64B and the multiplier 66B form the second soft limiter. In another embodiment of the present invention, the compression ratio may be less than infinite, and the threshold detection may be omitted, so that pure compression is applied rather than limiting.

In addition, event detection and control block 38 may be adapted to adapt adaptation of W (z) when one or both of the first and second limiters are active, and after the adaptive filter control equations are no longer applied, The W-coefficient control block 31 controls the adaptive filter 32 until all of the signal level detectors 64A and 64B indicate that the restriction is no longer applied to the noise- Lt; RTI ID = 0.0 > of the < / RTI >

5, a block diagram of an ANC system in accordance with an embodiment of the present invention that may be implemented in a CODEC integrated circuit 20 is shown. The reference microphone signal ref is generated by a delta-sigma ADC 41A, which operates at 64 times oversampling and whose output is applied to a factor of 2 through a decimator 42A To produce 32 times oversampling. The delta-sigma shaping device 43A distributes the energy of the images out of the bands where the final response of the parallel pair of adaptive filter stages 44A and 44B will have significant response. Filter stage (44B) is gatneunde a fixed response (W _FIXED (z)), such a fixed response (W _FIXED (z)) is typically P (z for a particular design of the wireless telephone 10 for the typical user ) / S (z). ≪ / RTI > The adaptive portion W _ADAPT (z) of the response of the estimate of P (z) / S (z) is provided by the adaptive filter stage 44A, which filters the leakage minimum-mean- And is controlled by the coefficient controller 54A. Leaked LMS coefficient controller 54A is normalized to a predetermined response over time during which the response is flat or otherwise not provided with any error input so that leakage LMS coefficient controller 54A is adapted so that leakage It is enemy. Providing a leakage controller prevents long-term instabilities that can occur under certain environmental conditions, and generally makes the system more robust against certain sensitivities of the ANC response.

3, the reference microphone signal ref is filtered by the filter 51 with the response SE _COPY (z) as the filter response SE (z), which is a replica of the estimate of the response of the path S (z) _COPY (z)), and the output of the filter 51 is decimated by a factor 32 through a decimator 52A to produce a baseband audio signal, which is an infinite impulse response IIR ) Filter 53A to the leakage LMS 54A. The error microphone signal err is generated by the delta-sigma ADC 41, which operates at 64 times oversampling and its output via decimator 42B by factor 2 Decimated to produce a 32 times oversampling signal. 3, the amount of downlink audio ds filtered by the adaptive filter to apply the estimated response of path S (z) is removed from error microphone signal err by combiner 46C, And the output of the combiner 46C is decimated by a factor 32 through a decimator 52C to produce a baseband audio signal and the baseband audio signal is passed through an infinite impulse response (IIR) (54A). The response S (z) is generated by another parallel set of adaptive filter stages 55A and 55B, one of which filter stage 55B has a fixed response SE _FIXED (z) Filter stage 55A has an adaptive response SE _ADAPT (z) that is controlled by leakage LMS coefficient controller 54B. The outputs of the adaptive filter stages 55A and 55B are combined by a combiner 46E. Similar to the implementation of the transfer function W (z) described above, the filter response SE _FIXED (z) generally provides a suitable starting point under various operating conditions for the electric / acoustic path S (z) Lt; / RTI > The independent control values are provided in the system of FIG. 5 to control the adaptive filter 51 with the response (SE _COPY (z)), shown as a single adaptive filter stage. However, the adaptive filter 51 may alternatively be used with two parallel stages, and the same control value used to control the adaptive filter stage 55A may then be used in the implementation of the adaptive filter 51, Can be used to control. The inputs of the leakage LMS control block 54B are also provided in baseband by decimating the downlink audio signal ds through a decimator 52B which decimator 52B is coupled by another combiner 46E Decimated by the factor 32 after the combiner 46C removes the signal generated from the combined outputs of the adaptive filter stage 55A and the filter stage 55B. The output of combiner 46C represents the error mic signal err from which the components due to the downlink audio signal ds have been removed and is provided to the LMS control block 54B after decimation by the decimator 52B. Another input of the LMS control block 54B is the baseband signal generated by the decimator 52C.

The above baseband and oversampled signaling devices provide simplified control and reduced power consumed in adaptive control blocks such as leakage LMS controllers 54A and 54B while at the same time providing adaptive filter stages 44A-44B , 55A-55B, and adaptive filter 51 at an oversampled rate. The remainder of the system of Figure 5 includes a combiner 46D that combines the downlink audio ds with a portion of the inner audio ia and near-end audio, with some of the near-end audio being generated by the sigma-delta ADC 41B And filtered by a lateral attenuator 56 to provide feedback conditions. The output of combiner 46D is shaped by sigma-delta forming machine 43B and the sigma-delta forming machine 43B provides inputs to filter stages 55A and 55B, 55A and 55B) are moved out of the band to have a significant response.

In accordance with one embodiment of the present invention, the output of combiner 46D is combined with the output of adaptive filter stages 44A-44B, which are processed by a control chain, A combiner 46A that combines the outputs of the mute blocks 45A and 45B and the hard mute blocks 45A and 45B and the noise canceling channel to ramp up or ramp down the gain of the noise- A soft mute 47, and a soft limiter 48 for generating a noise-free signal. The anti-noise signal is then subtracted by combiner 46B from the source audio output of combiner 46D. In this embodiment, the soft limiter 48 includes the speaker damage prevention circuit described above with reference to Figs. 3 and 4. Fig. The output of the combiner 46B is interpolated upward by the factor 2 through the interpolator 49 and then regenerated by the sigma-delta DAC 50 operating at a 64 times oversampling rate. The output of the DAC 50 is provided to an amplifier A1, which generates a signal to be transmitted to the speaker SPKR.

The event detection and control block 38 includes an output of the decimator 52C indicating how well the ANC system erases the acoustic noise measured at the error microphone E, Such as the output of the decimator 52A indicative of the environment, the downlink audio signal ds and the near-end audio signal ns. Depending on the detected acoustic events, or other environmental factors such as the position relative to the ear 5 of the wireless telephone 10, the event detection and control block 38 generates various outputs, 5, it is not shown in particular that the hard-mute blocks 45A-45B are applied, the characteristics of the mute 47 and the limiter 38, the leakage LMS control blocks 54A and 54B are stopped or reset, In some embodiments of the invention, it is possible to control which fixed responses are selected for the fixed portion of the adaptive filters, e.g., adaptive filter stages 44B and 55B.

Within the systems of FIG. 5, and / or within the exemplary circuits of FIGS. 2-4, each or a portion of the elements may be implemented directly in logic, or may be implemented in a manner that performs operations such as adaptive filtering and LMS coefficient calculations And may be implemented by a processor, such as a digital signal processing (DSP) core, that executes program instructions. Although the DAC and ADC stages are typically implemented with dedicated mixed-signal circuits, the architecture of the ANC system of the present invention is generally well suited to a hybrid approach, where such logic allows logic to be highly oversampled While the program code or microcode-driven processing elements are more complex, but may be used at a lower rate, such as computing tabs for adaptive filters and / or responding to detected events such as those described herein Lt; / RTI >

While the present invention has been particularly shown and described with reference to preferred embodiments thereof, those skilled in the art will recognize that the foregoing and other changes in form and detail may be made therein without departing from the spirit and scope of the invention.

Claims (28)

For a personal audio device,
A personal audio device housing;
A transducer mounted on said housing for reproducing an audio signal, said audio signal having both a source audio for reproduction to a listener and a noise-prevention signal for canceling the effects of ambient audio sounds in the acoustic output of said transducer Said transducer comprising;
A reference microphone mounted on the housing to provide a reference microphone signal representative of the ambient audio sounds;
An error microphone mounted on the housing to provide an error mic signal indicative of the acoustic output of the transducer; And
And a processing circuit in the housing for adaptively generating the noise-prevention signal from the reference microphone signal such that the noise-prevention signal causes a substantial erasure of the ambient audio sounds,
The processing circuit monitors the level of the noise-prevention signal, determines whether the noise-prevention signal can cause damage to the transducer, and determines whether the noise- And the processing circuit implements an adaptive filter having a response shaping the noise-free signal to reduce the presence of the ambient audio sounds in the error microphone signal, the processing circuit comprising: Stops adaptation of the adaptive filter in response to a determination that the adaptive filter may cause damage to the transducer. The method according to claim 1,
And the processing circuit limits or compresses the noise-protection signal in response to the determination that the noise-prevention signal has exceeded a first threshold value. 3. The method of claim 2,
Wherein the processing circuit first restricts or first compresses the noise-prevention signal in response to a determination that the noise-prevention signal has a low frequency component exceeding the first threshold value. The method of claim 3,
The processing circuit may limit or limit the first limited or first compressed result by determining that the full bandwidth of the result of first limiting or first compressing the signal exceeds a second threshold, Gt; compressed, < / RTI > personal audio device. delete The method according to claim 1,
The processing circuit is configured to first limit or first compress the noise-prevention signal in response to the determination that the noise-prevention signal has low frequency components exceeding a first threshold, Limiting or second compressing the first limited or first compressed result by determining that the full bandwidth of the result of limiting or first compressing exceeds the second threshold, And stops adaptation of the adaptive filter if low frequency components of the anti-noise signal exceed the first threshold. The method according to claim 6,
The processing circuit also stops adaptation of the adaptive filter if the full bandwidth of the result of first limiting or first compressing the signal exceeds a second threshold. The method according to claim 1,
The processing circuit is configured to first limit or first compress the noise-prevention signal in response to the determination that the noise-prevention signal has low frequency components exceeding a first threshold, Limiting or second compressing the first limited or first compressed result by determining that the full bandwidth of the limited or first compressed result exceeds the second threshold, And stops adaptation of the adaptive filter if either the first threshold or the second threshold is exceeded. The method according to claim 1,
Wherein the personal audio device further comprises a transceiver for receiving the source audio as a downlink audio signal. The method according to claim 1,
Wherein the personal audio device is an audio playback device and the source audio is a program audio signal. A method of preventing damage to a transducer of a personal audio device having adaptive noise cancellation,
Measuring ambient audio sounds through a reference microphone signal;
Adaptively generating a noise-free signal from a result of the measuring step to cancel influences of ambient audio sounds in the acoustic output of the transducer;
Combining the noise-prevention signal and the source audio signal;
Providing a result of said combining to a transducer;
Measuring the acoustic output of the transducer via an error microphone, wherein the adaptively generating comprises the step of measuring the acoustic output of the transducer to reduce the presence of the ambient audio sounds in the result of measuring the acoustic output of the transducer, - implementing an adaptive filter having a response shaping the anti-cancel signal; measuring the acoustic output;
Monitoring a level of the noise-prevention signal;
Determining if the noise-canceling signal can cause damage to the transducer;
Adjusting the noise-prevention signal so as to prevent damage to the transducer; And
And stopping the adaptation of the adaptive filter in response to a determination that the noise-canceling signal may cause damage to the transducer. 12. The method of claim 11,
Wherein said adjusting comprises restricting or compressing said noise-canceling signal in response to determining that said noise-canceling signal has exceeded a first threshold value. 13. The method of claim 12,
The limiting or compressing step includes first limiting or first compressing the noise-preventing signal in response to a determination that the noise-prevention signal has a low frequency component exceeding a first threshold value To prevent damage to the transducer. 14. The method of claim 13,
Limiting the result of the first constraining or first compressing step to the second constraint by determining that the complete bandwidth of the result of the first constraining or first compressing of the signal exceeds a second threshold, Further comprising compressing the first portion of the transducer to prevent damage to the transducer. delete 12. The method of claim 11,
Limiting or first compressing the noise-prevention signal in response to a determination that the noise-prevention signal has low frequency components exceeding a first threshold value;
Limiting the result of the first constraining or first compressing step to the second constraint by determining that the complete bandwidth of the result of the first constraining or first compressing of the signal exceeds a second threshold, Further comprising:
Wherein the stopping is performed in response to a determination that low frequency components of the noise-canceling signal exceed the first threshold. 17. The method of claim 16,
The step of stopping may also be performed in response to a determination that the complete bandwidth of the result of the first limiting or first compressing of the signal exceeds a second threshold value to prevent damage to the transducer Way. 12. The method of claim 11,
Limiting or first compressing the noise-prevention signal in response to a determination that the noise-prevention signal has low frequency components exceeding a first threshold value; And
Limiting the result of the first constraining or first compressing step to the second constraint by determining that the complete bandwidth of the result of the first constraining or first compressing of the signal exceeds a second threshold, Further comprising:
Wherein the stopping step is performed in response to a determination that the low frequency components of the noise-canceling signal exceed a first threshold, and wherein the step of stopping is performed using one of the first threshold value or the second threshold value Is exceeded, in order to prevent damage to the transducer. 12. The method of claim 11,
Wherein the personal audio device is a wireless telephone and the method further comprises receiving the source audio as a downlink audio signal. 12. The method of claim 11,
Wherein the personal audio device is an audio playback device and the source audio is a program audio signal. An integrated circuit for implementing at least a portion of a personal audio device,
An output for providing a signal to a transducer, said signal comprising both source audio for reproduction to a listener and a noise-prevention signal for canceling the effects of ambient audio sounds in the acoustic output of said transducer, ;
A reference microphone input for receiving a reference microphone signal representative of said ambient audio sounds;
An error microphone for receiving an error mic signal indicative of the acoustic output of the transducer; And
And a processing circuit for adaptively generating the noise-prevention signal from the reference microphone signal such that the noise-prevention signal causes a substantial erasure of the ambient audio sounds,
The processing circuitry is further configured to monitor the level of the noise-avoiding signal and to determine whether the noise-canceling signal can cause damage to the transducer, to prevent damage to the transducer, Prevention signal, the processing circuit implementing an adaptive filter having a response shaping the noise-canceling signal to reduce the presence of the ambient audio sounds in the error microphone signal, the processing circuit comprising: - stop the adaptation of the adaptive filter in response to a determination that an anti-cancel signal may cause damage to the transducer. 22. The method of claim 21,
Wherein the processing circuitry limits or compresses the noise-protection signal in response to determining that the noise-prevention signal has exceeded a first threshold. 23. The method of claim 22,
Wherein the processing circuit first restricts or first compresses the noise-prevention signal in response to a determination that the noise-prevention signal has a low frequency component that exceeds a first threshold value. 24. The method of claim 23,
The processing circuit may be configured to limit or suppress the first limited or first compression result by determining that the complete bandwidth of the result of first limiting or first compressing the signal exceeds a second threshold, Th integrated circuit. delete 22. The method of claim 21,
The processing circuit is configured to first limit or first compress the noise-prevention signal in response to the determination that the noise-prevention signal has low frequency components exceeding a first threshold, Limiting or second compressing the first limited or first compressed result by determining that the full bandwidth of the limiting or first compressing result exceeds the second threshold, And stops the adaptation of the adaptive filter if the low frequency components of the anti-noise signal exceed the first threshold. 27. The method of claim 26,
The processing circuit also stops adaptation of the adaptive filter if the complete bandwidth of the result of first limiting or first compressing the signal exceeds the second threshold. 22. The method of claim 21,
The processing circuit is configured to first limit or first compress the noise-prevention signal in response to the determination that the noise-prevention signal has low frequency components exceeding a first threshold, Limiting or second compressing the first limited or first compressed result by determining that the full bandwidth of the limiting or first compressing result exceeds the second threshold, And stops adaptation of the adaptive filter if either the first threshold or the second threshold is exceeded.

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